2 Copyright (C) 2021 Carl Hetherington <cth@carlh.net>
4 This file is part of DCP-o-matic.
6 DCP-o-matic is free software; you can redistribute it and/or modify
7 it under the terms of the GNU General Public License as published by
8 the Free Software Foundation; either version 2 of the License, or
9 (at your option) any later version.
11 DCP-o-matic is distributed in the hope that it will be useful,
12 but WITHOUT ANY WARRANTY; without even the implied warranty of
13 MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the
14 GNU General Public License for more details.
16 You should have received a copy of the GNU General Public License
17 along with DCP-o-matic. If not, see <http://www.gnu.org/licenses/>.
22 #include "audio_analyser.h"
23 #include "audio_analysis.h"
24 #include "audio_buffers.h"
25 #include "audio_content.h"
26 #include "audio_filter_graph.h"
27 #include "audio_point.h"
29 #include "dcpomatic_log.h"
33 #include <dcp/warnings.h>
36 LIBDCP_DISABLE_WARNINGS
37 #include <libavutil/channel_layout.h>
38 #ifdef DCPOMATIC_HAVE_EBUR128_PATCHED_FFMPEG
39 #include <libavfilter/f_ebur128.h>
41 LIBDCP_ENABLE_WARNINGS
45 using std::make_shared;
47 using std::shared_ptr;
49 using namespace dcpomatic;
52 static auto constexpr num_points = 1024;
55 AudioAnalyser::AudioAnalyser (shared_ptr<const Film> film, shared_ptr<const Playlist> playlist, bool from_zero, std::function<void (float)> set_progress)
57 , _playlist (playlist)
58 , _set_progress (set_progress)
59 #ifdef DCPOMATIC_HAVE_EBUR128_PATCHED_FFMPEG
60 , _ebur128(film->audio_frame_rate(), film->audio_channels())
62 , _sample_peak (film->audio_channels())
63 , _sample_peak_frame (film->audio_channels())
64 , _analysis (film->audio_channels())
67 #ifdef DCPOMATIC_HAVE_EBUR128_PATCHED_FFMPEG
68 _filters.push_back({"ebur128", "ebur128", "audio", "ebur128=peak=true"});
69 _ebur128.setup(_filters);
72 _current = std::vector<AudioPoint>(_film->audio_channels());
75 _start = _playlist->start().get_value_or(DCPTime());
78 for (int i = 0; i < film->audio_channels(); ++i) {
80 _sample_peak_frame[i] = 0;
83 auto add_if_required = [](vector<double>& v, size_t i, double db) {
85 v[i] = pow(10, db / 20);
89 auto content = _playlist->content();
90 if (content.size() == 1 && content[0]->audio) {
92 for (auto channel: content[0]->audio->mapping().mapped_output_channels()) {
93 /* This means that if, for example, a file only maps C we will
94 * calculate LEQ(m) for L, R and C. I'm not sure if this is
97 _leqm_channels = std::min(film->audio_channels(), channel + 1);
100 _leqm_channels = film->audio_channels();
103 /* XXX: is this right? Especially for more than 5.1? */
104 vector<double> channel_corrections(_leqm_channels, 1);
105 add_if_required (channel_corrections, 4, -3); // Ls
106 add_if_required (channel_corrections, 5, -3); // Rs
107 add_if_required (channel_corrections, 6, -144); // HI
108 add_if_required (channel_corrections, 7, -144); // VI
109 add_if_required (channel_corrections, 8, -3); // Lc
110 add_if_required (channel_corrections, 9, -3); // Rc
111 add_if_required (channel_corrections, 10, -3); // Lc
112 add_if_required (channel_corrections, 11, -3); // Rc
113 add_if_required (channel_corrections, 12, -144); // DBox
114 add_if_required (channel_corrections, 13, -144); // Sync
115 add_if_required (channel_corrections, 14, -144); // Sign Language
116 add_if_required (channel_corrections, 15, -144); // Unused
118 _leqm.reset(new leqm_nrt::Calculator(
120 film->audio_frame_rate(),
123 850, // suggested by leqm_nrt CLI source
124 64, // suggested by leqm_nrt CLI source
125 boost::thread::hardware_concurrency()
128 DCPTime const length = _playlist->length (_film);
130 Frame const len = DCPTime (length - _start).frames_round (film->audio_frame_rate());
131 _samples_per_point = max (int64_t (1), len / num_points);
136 AudioAnalyser::analyse (shared_ptr<AudioBuffers> b, DCPTime time)
138 LOG_DEBUG_AUDIO_ANALYSIS("AudioAnalyser received %1 frames at %2", b->frames(), to_string(time));
139 DCPOMATIC_ASSERT (time >= _start);
140 /* In bug #2364 we had a lot of frames arriving here (~47s worth) which
141 * caused an OOM error on Windows. Check for the number of frames being
142 * reasonable here to make sure we catch this if it happens again.
144 DCPOMATIC_ASSERT(b->frames() < 480000);
146 #ifdef DCPOMATIC_HAVE_EBUR128_PATCHED_FFMPEG
147 if (Config::instance()->analyse_ebur128 ()) {
152 int const frames = b->frames ();
153 vector<double> interleaved(frames * _leqm_channels);
155 for (int j = 0; j < _leqm_channels; ++j) {
156 float const* data = b->data(j);
157 for (int i = 0; i < frames; ++i) {
160 interleaved[i * _leqm_channels + j] = s;
162 float as = fabsf (s);
164 /* We may struggle to serialise and recover inf or -inf, so prevent such
165 values by replacing with this (140dB down) */
168 _current[j][AudioPoint::RMS] += pow (s, 2);
169 _current[j][AudioPoint::PEAK] = max (_current[j][AudioPoint::PEAK], as);
171 if (as > _sample_peak[j]) {
172 _sample_peak[j] = as;
173 _sample_peak_frame[j] = _done + i;
176 if (((_done + i) % _samples_per_point) == 0) {
177 _current[j][AudioPoint::RMS] = sqrt (_current[j][AudioPoint::RMS] / _samples_per_point);
178 _analysis.add_point (j, _current[j]);
179 _current[j] = AudioPoint ();
184 _leqm->add(interleaved);
188 DCPTime const length = _playlist->length (_film);
189 _set_progress ((time.seconds() - _start.seconds()) / (length.seconds() - _start.seconds()));
190 LOG_DEBUG_AUDIO_ANALYSIS_NC("Frames processed");
195 AudioAnalyser::finish ()
197 vector<AudioAnalysis::PeakTime> sample_peak;
198 for (int i = 0; i < _film->audio_channels(); ++i) {
199 sample_peak.push_back (
200 AudioAnalysis::PeakTime (_sample_peak[i], DCPTime::from_frames (_sample_peak_frame[i], _film->audio_frame_rate ()))
203 _analysis.set_sample_peak (sample_peak);
205 #ifdef DCPOMATIC_HAVE_EBUR128_PATCHED_FFMPEG
206 if (Config::instance()->analyse_ebur128 ()) {
207 void* eb = _ebur128.get("Parsed_ebur128_0")->priv;
208 vector<float> true_peak;
209 for (int i = 0; i < _film->audio_channels(); ++i) {
210 true_peak.push_back (av_ebur128_get_true_peaks(eb)[i]);
212 _analysis.set_true_peak (true_peak);
213 _analysis.set_integrated_loudness (av_ebur128_get_integrated_loudness(eb));
214 _analysis.set_loudness_range (av_ebur128_get_loudness_range(eb));
218 if (_playlist->content().size() == 1) {
219 /* If there was only one piece of content in this analysis we may later need to know what its
220 gain was when we analysed it.
222 if (auto ac = _playlist->content().front()->audio) {
223 _analysis.set_analysis_gain (ac->gain());
227 _analysis.set_samples_per_point (_samples_per_point);
228 _analysis.set_sample_rate (_film->audio_frame_rate ());
229 _analysis.set_leqm (_leqm->leq_m());